1. Field of the Invention
The present invention relates to an internet telephone communication system, and more particularly, to an internet telephone communication system that improves a VOIP conversation quality by varying a number of retransmissions based on the data loss rate of the voice data packets received.
2. Background of the Related Art
In a telephone communication system using an internet network, analog voice signals are initially digitalized, encoded, and get transmitted in forms of voice data packets by a transmitter. A receiver then receives the voice data packets sent, converts them to analog signals, and outputs the analog signals through a speaker.
FIG. 1 illustrates a typical internet telephone communication system. According to FIG. 1, the system includes a voice transmitting part that digitalizes and encodes a voice signal and sends the encoded voice data in forms of voice data packets through an internet network; and a voice receiving part that receives the voice data packets and restores the voice data so that the data can be heard. First, the voice transmitting part includes a microphone 101 for receiving a voice signal; an ADC (analog to digital converter) 102 for digitalizing the analog voice signal received from the microphone 101; a voice coder 103 for compressively encoding the digitalized voice signal; and a transmitting protocol processor 104 for generating voice data packets by processing the compressed voice data in accordance to an internet protocol and sending the voice data packets through an internet network 105. The voice receiving part includes a receiving protocol processor 106 for receiving the voice packets transmitted through an internet network 105 and extracting the compressed voice data; a voice decoder 107 for restoring the compressed voice data; a DAC (digital to analog converter) 108 for converting the stored data into analog signals; and a speaker 109 for outputting the converted analog signals.
The system illustrated in FIG. 1 works as follows. First of all, a voice signal generated from a sender is inputted to the microphone 101, and the ADC 102 converts the signal to digital voice data. After the voice coder 103 compressively encodes the digitalized voice data to increase the transmission efficiency, supplemental data such as header and trailer data are added to the compressed voice data in the transmitting protocol processor 104, and the data get transmitted through the internet network 105 in forms of the voice data packets. Then the protocol processor 106 of the receiving part receives the transmitted data, eliminates the supplemental data (i.e., header and trailer) and extracts the compressed voice data. The extracted voice data are then restored in the voice decoder 107, and the DAC 108 converts the restored digital voice data to an analog signal. Finally, the speaker 109 outputs the converted analog voice signal.
However, the internet phone system described above has a problem that the VOIP conversation quality is drastically decreased when some of the voice packets are lost during the transmission or signal process steps. When they are lost in the internet network, empty spaces will be included instead in the received voice data, and the telephone receiver will hear discontinuous sound so that it will be difficult to understand what the caller says. Therefore, since the system does not have any mechanism to prevent such voice packet losses or to correct the error caused by such losses, such losses result a poor VOIP conversation quality.